Hi, I've been playing a lot with bristol synth and really love it. So
much so that I've been trying to 'Jackify' it. Actually, I'm pretty
much done, but can't figure out the internal audio format. Its
interleaved floats I think, but not normalised to [-1,1]. If any of
the developers are here could you help me out? I can hear noise, but I
need to tune the maths. TIA.
--ant
>> That said, I think Patrick is right to start thinking about this now.
Thanks.
>I think he's completely right - I'm not sure about this bank account
>thing but I do think that now is the time to be demoing, talking up and
>generally approaching people and companies about Linux music software.
>I wrote us up (and mentioned a few other apps) in the latest edition of
>Linux User - John at mstation.org has been very kind so far as well.
>Now is the right time to be talking to people and getting the
>"products" out there. If it works - why not tell people about it?
The reason I believe we need to have various bank accounts are because
we cannot afford to waste money on excessive service charges and not
everyone has access to credit cards. If we have the accounts in the
right countries then people can just donate cash.
From a professional perspective we need to show our prospective clients
that we have sound financial thinking. It's mostly a subconscious need
that consumers have. They want to know that the money they are investing
is being given to people/companies/organisations who use it. Most people
don't really care how it is used although we have the moral
justification on our side too.
This is from the Sound on Sound advertising package.
"The main target market of Sound On Sound is the professional
and semi-professional musician who is the kind of person that will have
the spending ability to purchase a large range of products from
synthesizers to samplers, mixing desks to microphones, multitracks to
monitors, effects to expanders and computer hardware and software.
They are not time wasters who do not know their profession - they are
serious and mature individuals working with a reasonable budget."
If we want to appeal to this audience we need to prove to them that they
are investing in professional audio. We need to wine them and dine them
(metaphorically). If they look into our commmunity and say these are
just amateur geeks who have made some interesting things happen it won't
work. If we take the intiative and lead them into our world they will
come at it from the perspective that we are professionals who have
created a very credible concept that we are proud of and want them to
enjoy using.
They will ask "What kind of cash have we invested" and if we come back
with "Ahh, well we don't actually have a scope on the financial side of
our open community." They are just going to look around for a while and
leave.
If we can show them that not only are we mathematics and logics wizards
but that we also have solid business sense then they are going to stick
around and see what we have to offer. A lot of them will probably invest
just to test the waters or to keep up with the play.
I want to see an advertising campaign happen that will educate and
encourage the mass of potential user to take the step. I also want to
make sure that we have covered our asses when they finally walk in
through the doors.
It's a choice between being amatuer enthusiasts or professionals.
If we come across as professionals people won't give a toss about
geekyness.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem
The world and his dog seems to be releasing macos/windows audio s/w that
looks like 19" rack units.
Anyone know enough about X to know if its possible to make X apps open
thier main window inside a standard sized cabinet (ala Reason).
I'm assuming it would be ok to require the app to be a certian size and
have explicit support, but I guess it couldn't put any restrictions on
toolkit.
Other than looking cool, it would actually be a useful way to keep window
clutter down. Not that we have any 19" lookalike apps yet, but I guess we
will do at some point.
- Steve
this is not meant to intimidate, rather to be a wake-up call.
it seems almost unreal (and certainly unprofessional) to me
that an instrument plugin api is being discussed here by a
bunch of people who have little to no experience in the field
of software sequencers. going into implementation details at
the current level of understanding of the problem space is,
excuse me, ridiculous.
after all, what is going to drive your instrument networks?
punched cardboard? certainly not. you'll either use realtime
input, or a sequencer, or, most wanted, a combination of both.
the closer the integration of the event/plugin system with the
sequencer, the more uses the api can be put to, with less pain.
stopping short of the mark where the api becomes useful for
more applications than basically sample-rate dependent
event->audio converters is narrow-minded. viewing the 'host'
as a blackbox supposed to 'do the rest' without caring about
its internals is blatant ignorance.
i do think it's reasonable to ignore my personal input since
i don't offer published code to back up my views, and when i
do, you'll find it centered around my personal musical needs.
however there are, afaik, people from the rosegarden team on
this list. it would also be helpful having werner schweer of
muse fame participate in some way or other. you might also want
to look at other free/open sequencer engines. for one thing,
you'll find that most, if not all, are tick-based.
vst[i] is a bad candidate i think because few people here will
have vst host-side coding experience, and the api itself is
bound to be centered around the particular coding needs of a
specific company, for a specific application that drags code
with it that originated in the eighties and never was subject
to public source-level review.
in short, the more people with hands-on sequencer experience
participating, the better. none are just too few.
tim
ps: if this post hasn't substantially changed your ways of
perceiving this matter, please don't bother answering.
[sorry for the wide distribution]
Is anyone out there trying out this combination?:
kernel 2.4.20-rc1
capabilities patch
low latency patch
preemptible kernel patch
almost current alsa cvs [20021028.170432 usa pacific time]
If I start jackd and then use the freqtweak jack client I get a completely
dead machine in a very short time (from a few seconds to 10 or 20
seconds).
Same alsa driver, but running on 2.4.19 (up to 20-pre4) seems to be fine
and I can play around with freqtweak for as long as I want :-)
No clues are left behind... something in the kernel is deadlocking, I
guess. I tried removing first the preemptible kernel patch with the same
result. Just a moment ago I tried again after removing the low latency
patch and I still get the same result... that pretty much leaves alsa and
the kernel. Got the same result on a laptop intel810 and an ice1712 card.
Interestingly enough just the jack server running is not enough to kill
the machine. Adding a jack client (just tried alsaplayer, same problem as
freqtweak) kills the machine really fast.
Maybe there is a problem with the scheduler when there are several
SCHED_FIFO tasks competing for the processor?
-- Fernando
I don't know whether this is a bug in ams, or a problem with my
system...I suspect the later, but can't figure out what's wrong.
But, I am unable to get ams to run. It appears to compile OK. But, as
noted below, running it results in a "Segmentation fault".
I have installed qt3 and fftw. I edited the top of make_ams to match my
system:
box1:~/audio_code/ams-1.5.5$ head -5 make_ams
QT_BASE_DIR=/usr/lib/qt3
QT_LIB_DIR=$(QT_BASE_DIR)/plugins-mt/styles
QT_BIN_DIR=/usr/bin
QT_INCLUDE_DIR=/usr/include/qt
X11_LIB_DIR=/usr/X11R6/lib
My ditribution is Debian stable (woody). My system is a PII 400Mhz
w/768MB RAM.
The version of g++ is 2.95.4.
I have a working install of jackd 0.44.0 from cvs. I'm starting jackd
like so:
# jackd -v -R -d alsa -d ymfpci -r 44100 -zt -n 3 -p 512
I can start up multiple instances of jack_metro, connect them to the
alsa ports and hear the beeps. So, jack appears to be working.
But, if I start ams nothing is reported in the jackd output about a new
client.
ams does attempt to connect to the X server when started. I know this
because if I 'su -' to get to root and then start ams:
box1:/home/eric/audio_code/ams-1.5.5# ./ams
ams: cannot connect to X server
but, if I su without the dash so that root inherits my user environment
ams gets past the point of connecting to the X server (assuming that
comes first) and exits with a segmentation fault.
I'm not sure what other information would be useful to you in helping me
to sort this out. If there is anything else I should tell you, please
let me know.
Thanks in advance,
Eric Rz.
-------- Original Message --------
Subject: ams segmentation fault
Date: Sat, 28 Dec 2002 21:47:34 -0500
From: Eric Dantan Rzewnicki <eric(a)zhevny.com>
Organization: zhevnycom
To: Eric Dantan Rzewnicki <rzewnickie(a)rfa.org>
CC: linux-audio-user(a)music.columbia.edu
References: <3E0BCBEC.9DB82093(a)zhevny.com>
<20021227103148.GH681(a)ecs.soton.ac.uk>
<1041005858.4921.17.camel@eviltwin> <20021227181307.GD19938(a)rfa.org>
Eric Dantan Rzewnicki wrote:
> I can now start jackd as root and
> then start ams -j. Sorry I didn't think of this earlier. :-\
>
> Adding /usr/local/lib to /etc/ld.so.conf and running ldconfig as Steve
> suggested allows ams to find libjack.so.0 (I really should have
> remembered that piece, too).
>
> Now that I have it running I'll go finish reading the ams documentation
> and figure out how to use it. :)
hmm... I perhaps spoke too soon.
I had ams working yesterday on one box (at the office)... But, now I
can't seem to get it working at home. It seems to compile fine, but when
i try to start it:
box1:/home/eric/audio_code/ams-1.5.5# ./ams
Segmentation fault
box1:/home/eric/audio_code/ams-1.5.5# ./ams -j
Segmentation fault
It doesn't matter if jackd is running or not. Either way it seg
faults...
What could cause this?
What information should I provide to help sort this out?
Thanks,
Eric Rz.
I have made a program which is to be run on a dedicated computer,
equipped only with a multichannel soundcard (hammerfall), a small LCD
and a remote control. This forms a dedicated surround processor (if
interested in the software, look at
http://www.ludd.luth.se/users/torger/almusvcu.html) for use in
multichannel HiFi systems.
In some configurations, a 64 sample processing block is relevant,
leaving a total IO-delay of 128 samples, which in 44.1 kHz is about 3
ms.
With such a configuration, I have got the system going for a few hours
at most before underflow. Of course the relevant processes are realtime
scheduled, I have even used tricks with sched_yield to get the
processes yield in the right order (I will probably have to pay for
that sooner or later though) for the most safe operation.
So, my question to this list is if someone has run a similar system for
days without underflows, and if there is some guide or tips on how to
put together the for the moment best low-latency kernel, if there are
kernel features that should be avoided, some bad driver or filesystem
perhaps, and if there are some hardware setting tricks (apart from
giving the sound card the highest interrupt priority). Perhaps tips on
which user space processes that should not run (I have no syslogd or
cron, but sshd is running)
Is it even possible today to build an embedded system with 1.5 ms
processing block based on Linux, which runs until hardware or power
failure? Oh, my software does not necessarily need to access any
filesystem while it is processing, although it would be good if it
could (to continuously save volume settings and similar if changed
through the remote control). So it can be seen as a strict
soundcard-to-the-cpu-and-back problem.
/Anders Torger
Hi,
Autocomp is an automatic accompaniment generator for Perl and Csound,
developed on Linux. It's an attempt at a practice utility for musicians.
Freshmeat project page is here:
http://freshmeat.net/projects/autocomp/
Download it or read more here:
http://www.waz.easynet.co.uk/software.html
Currently there is only one style of output, which is a pastiche of a
'swing' style with a walking bassline. The output is heavily on the
gorgonzola side of cheesy, but it's already at a stage where you can
practice with it if you don't mind the cheese level.
All comments/suggestions etc gratefully received.
Cheers etc.,
Wayne
--
Wayne Myers
http://www.waz.easynet.co.uk/
>Actually I'm just asking because I want to see how good jack is and
how >it works before we start to add jack support to glame.
Works well for me.
Having jack support increases the liklihood of more people using glame too.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
Being on stage with the band in front of crowds shouting, "Get off! No!
We want normal music!", I think that was more like acting than anything
I've ever done.
Goldie, 8 Nov, 2002
The Scotsman
Mmm.. releases :)
Biggest thing is that it now listens to changes in alsa and
automatically refreshes.
* made a nice dlopen() system for GUIs and drivers
* ladcca support
* major amounts of code cleaning up and seperating (eg apb.h -> 6 .h's:)
* bug fixes
* compilation fixes
* listens to alsa changes in a thread
* gtk gui now uses a proper toolbar with stock buttons
http://pkl.net/~node/alsa-patch-bay.html
Bob